Internet comprises an enormous number of different networks interconnected by computers. Internetworking implies that the interconnected systems agree to conventions that allow each computer to communicate with every other computer. In particular, an internet will allow two machines to communicate even if the communication path between them passes across a network to which neither connects directly. Such cooperation is only possible when computers agree on a set of universal identifiers and a set of procedures for moving data to its final destination.
In an internet, interconnections among networks are formed by computers called IP routers, or IP gateways, that attach to two or more networks. A gateway forwards packets between networks by receiving them from one network and sending them to another.
Data communications are often conducted over data-based networks, such as those based on the Internet Protocol. Asynchronous data transfer technologies, like ATM (Asynchronous Transfer Mode), is commonly used as layer 2. ATM is a connection-oriented network technology that uses small, fixed-size cells at the lowest layer. It has the potential advantage of being able to support voice, video and data with a single underlying technology.
However, the Internet is quickly changing from an asynchronous data transfer conduit to a multimedia backbone. Continuous media applications, such as audio or video conferencing, Internet radio, on-line seminars, or Video-on-Demand system, are becoming commonplace. These applications transmit data at regular intervals and require strict guarantees on maximum delay and minimum bandwidth. Users accustomed to the high bandwidth provided by fixed network connections as well as other multimedia services. As they move to wireless devices, they will expect similar services even when connected via low-bandwidth wireless connections.
At the transport layer, the Real Time Transport Protocol (RTP) is used to support multimedia traffic on the Internet. Some of the benefits of using RTP are that it does not require changes to existing routers and gateways, it may be implemented on top of the User Datagram Protocol (UDP/IP) or ATM, and it takes advantage of the multimedia backbone, which allows bandwidth-efficient distribution of data to many users by eliminating redundant packet transmissions.
RTP has been designed within the Internet Engineering Task Force (IETF). Note that the moniker “transport protocol” could be misleading, as it is currently mostly used together with UDP, also designated as a transport protocol. RTP is an end-to-end protocol. RTP consists of two parts, a data part and a control part. Continuous media data like audio and video is carried in RTP data packets.
RTP offers a control protocol called RTCP that supports the protocol functionality. An RTP message consists of a number of “stackable” packets, each with its own type code and length indication. Their format is fairly similar to data packets; in particular, the type indication is at the same location. RTCP packets are multicast periodically to the same multicast group as data packets. Thus, they also serve as a liveness indicator of session members, even in absence of transmitting media data.
RTCP packets contain the necessary information for Quality-of-Service (QoS) monitoring. If they are multicast, all session members can survey how the other participants are faring. Applications that have recently sent audio or video data generate a sender report. It contains information useful for intermedia synchronisation as well as cumulative counters for packets and bytes sent. These allow receivers to estimate the actual data rate. Session members issue receiver reports for all video and data audio sources they have heard from recently. They contain information on the highest sequence number received, the number of packets lost, a measure of the interarrival jitter and timestamps needed to compute an estimate of the roundtrip delay between sender and the receiver issuing the report.
The receiving end applications deliver Receiver Reports to the source. The reports include information that enables the calculation of packet losses and packet delay jitter. There are two reasons for packet loss: packets get lost due to buffer overflow or to bit errors. The probability of bit errors is very low on most networks. It is already considered in the TCP protocol that loss is rather induced by congestion than by bit errors. Buffer overflow can happen on a congested link or at the network interface of the end device.
Document WO 0079830, A1, discloses a telecommunication system giving the option to a user (A1) to select desired services delivered via a service network from service providers. One of the problem addressed in this document is how to guarantee the services a transmission quality. This is solved by giving the services a priority corresponding to the need they have for transmission in real time. Two types of priority fields in the data packets are discussed: CoS, i.e. Class of Service, which is a field in the link level (layer 4), and ToS, i.e. Type of Service, which is a field in the Internet level (layer 3). The priority method is discussed in more detail from line 6 of page 10 to line 34 of page 13.
Document WO 0131969, A1, discloses an Ethernet-type edge switch that interfaces a cell-based (ATM) network, which transport multimedia information. A system providing end-to-end Quality of Service in ATM systems is presented. A CIF shim layer is employed to manage the transmission of all traffic according to the Quality of Service specified for each traffic stream (see line 20 of page 18-line 20 of page 19).
Document WO 0003521, A1, discloses a middleware-based real-time communication system. The nodes of the present Ethernet network comprises an application layer, a Middleware Real Time Ethernet (MRTE) layer and an Ethernet Protocol layer. The MRTE layer logically comprises a pair of queues for data traffic or packets generated in the application layer for transmission to another node. The first queue comprises a real time queue for queuing information packets that have been accepted for transmission on a real time basis. The real time traffic has been sorted by criticality and is guaranteed to be sent without collision. The second queue comprises a non-real time queue for data packets that do not need to arrive at a destination in real time to be of value to the receiving node. This queue is sorted by first in, first out. The MRTE layer is further divided into QoS adaptation services and deterministic scheduling services. The QoS adaptation services contains a QoS manager and a QoS adaption algorithm.
The QoS manager and its associated QoS adaptation algorithm provide QoS-based negotiation and adaptation services such as changing the duration of non-real-time data traffic or suspending low-criticality traffic to ensure that sufficient collision free bandwidth is provided for high priority real time traffic. Deterministic scheduling services contains a collision resolution protocol, an MRTE protocol and an MRTE scheduler and its associated scheduling algorithm and MRTE repository. The deterministic scheduling algorithm controls the flow of communication with the QoS manager, which further controls the flow of communication with the applications and the QoS adaption algorithm (see line 34 of page 7 to line 21 of page 8 and line 24 of page 10 to line 15 of page 14).
Document EP 1 146 704, A2, discloses a system and method for providing an intermediary layer for VOIP call pipe establishment. A Generate QoSEthernet layer is provided, interposed between an IP protocol voice communication layer and the QoSEthernet layer. The Generate QoSEthernet layer intercepts call commands, such as set up call commands, and identifies a required QoS for the particular call (see section [0026]).
Document WO 9965196, A1, discloses a local area network (LAN) comprising a communication switching module. Said module controls flow of both delay-sensitive voice digital voice signals from digital telephones and non-delay-sensitive user data from PC's. The module is connected to a WAN. The Ethernet LAN employs a Constant Bit Rate Channel.
Numerous problems may occur for many applications, when QoS is not controlled.
One problem, for example, will arise when the media stream arrives to a network not supporting the QoS feature. The switches of the network may be of Ethernet Layer 2 kind or ATM offering connection oriented service. There is no possibility for giving selective priority, like precedence, to different IP data packages. Quality of Service (QoS) on IP-level layer 3 is therefore not provided for.
A solution is needed when applications based on IP require priority, due to Quality of Service, from a WAN based on a narrow band network access technology like Cable TV (CaTV), xDSL (various DSL systems [Digital Subscriber Line], e.g. Asymmetric DSL), or Wireless Networks like Wireless Local Area Networks (WLAN), 802.11b or Fixed Wireless Access (FWA) technology.